There has been known a method of correcting the sound field in the listening environment by filtering an acoustic signal outputted from a speaker or the like. At the present time, an infinite impulse response (IIR) filter and finite impulse response (FIR) filter are widely known as digital filters used in filtering.
If the gain of a desired frequency band is set in order to filter a signal using an IIR filter, the set gain of the frequency band can be relatively easily adjusted. However, an IIR filter performs a feedback loop process as an internal process and therefore accumulates arithmetic errors, deteriorating the arithmetic accuracy. On the other hand, an FIR filter performs one multiplication and one addition on each delayed signal for each coefficient of a filter as an internal process and therefore is characterized in that it does not accumulate arithmetic errors and has high arithmetic accuracy. However, in the case of an FIR filter, unlike an IIR filter, it is difficult to generate a filter by partially changing the frequency characteristics of the FIR filter (by changing a particular portion of the frequency characteristics). For this reason, even if the gain of a desired frequency band is set, it is not easy to adjust the set gain of the frequency band using an FIR filter. Accordingly, an FIR filter has a problem of having less adjustability than an IIR filter.
To solve this problem, there have been developed signal processing circuits that have both the adjustability of an IIR filter and the high accuracy of an FIR filter, which accumulates less arithmetic errors (for example, see Patent Literature 1).